[Asterisk-Users] Realtime SIP Registrations

[Douglas Garstang]

I'm intensely curious why it doesn't currently work.
You have multiple Asterisk systems, all referring to a common table for SIP peer information. 
The fact that there is multiple Asterisk systems accessing the same MySQL data should be completely transparent to each of them, and I don't understand why this doesn't work.

Anyone?

Doug.

> -----Original Message-----
> From: David Thomas [mailto:punknow at gmail.com]
> Sent: Friday, June 30, 2006 9:40 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Realtime SIP Registrations
> 
> 
> Doug,
> 
> If you'd be willing to share the patch and AGI, I would be happy to
> help test your solution. I know that myself and several others have
> been looking for a way to make Asterisk do this for quite some time.
> 
> regards,
> David
> 
> On 6/29/06, Doug G <Asterisk at isgcom.com> wrote:
> > Well, to dial a peer direclty the only thing that is 
> missing in realtime is the status of the sip peer.  
> (registered, Unregistered, unknown, reachable).   If you dial 
> a peer via ip and it is unavaliable you get dead air.  So you 
> need to know the status of the peer before dialing it.   The 
> change basicly updates realtime with the peers status.  I did 
> the same thing for IAX as well..
> >
> > Doug
> >
> >
> > ________________________________
> >
> > From: asterisk-users-bounces at lists.digium.com on behalf of 
> Mike Lynchfield
> > Sent: Thu 6/29/2006 1:43 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Realtime SIP Registrations
> >
> >
> > can you elaborate on modify sip to update the "status" on 
> the sip friends in realtime
> > thanks
> >
> >
> > On 6/29/06, Doug G < Asterisk at isgcom.com 
> <mailto:Asterisk at isgcom.com> > wrote:
> >
> >        What I did was modify sip to update the "status" on 
> the sip friends in realtime.   Then via FAGI dial them 
> directly with the data found in real-time. (ie dial ( 
> SIP/1112223333 at 10.10.10.1:5060) Of course you need to check 
> the "status" in realtime data before you dial.  This allows 
> MANY Asterisk servers to share the same SIP data.    I then 
> load balance with DNS SRV..  Yes I have tested in failover it works.
> >
> >
> >
> >        I too have been told that by many that this will not 
> work.  So I keep expecting to hit some problem with it, but 
> to date I have not...
> >
> >
> >
> >        Doug
> >
> >
> >
> >
> >
> >        ________________________________
> >
> >        From: asterisk-users-bounces at lists.digium.com on 
> behalf of David Thomas
> >        Sent: Thu 6/29/2006 1:05 PM
> >        To: Asterisk Users Mailing List - Non-Commercial Discussion
> >        Subject: Re: [Asterisk-Users] Realtime SIP Registrations
> >
> >
> >
> >        I think lots of us know about it... We're just not 
> sure how to go
> >        about fixing it. :-(
> >        I know it's been a thorn in my side since I started 
> using Asterisk.
> >
> >        I would suspect that many of those saying "works for 
> me" have never
> >        actually tested their system in failure scenarios, 
> or they are working
> >        in a controlled environment without NAT and such...
> >
> >        regards,
> >        David
> >
> >        On 6/29/06, Douglas Garstang <dgarstang at oneeighty.com> wrote:
> >        > > -----Original Message-----
> >        > > From: Aaron Daniel [mailto: amdtech at shsu.edu 
> <mailto:amdtech at shsu.edu> ]
> >        > > Sent: Thursday, June 29, 2006 9:27 AM
> >        > > To: Asterisk Users Mailing List - Non-Commercial 
> Discussion
> >        > > Subject: RE: [Asterisk-Users] Realtime SIP Registrations
> >        > >
> >        > >
> >        > > On Thu, 2006-06-29 at 09:15 -0600, Douglas 
> Garstang wrote:
> >        > > > How about fixing realtime SIP so that multiple Asterisk
> >        > > boxes can reference the same database?
> >        > > >
> >        > > > Doug.
> >        > >
> >        > > That's kinda what I'm hoping to work towards :)
> >        >
> >        > I'm surprised you even knew about that. There 
> seems to be a common misconception that this should work 
> (caused by common sense maybe). Every time I bring it up, 
> people go 'Of course it works!', or 'Works for me!' (still 
> don't know why it works for some and not others.....)
> >        >
> >        > Doug.
> >        > _______________________________________________
> >        > --Bandwidth and Colocation provided by Easynews.com --
> >        >
> >        > Asterisk-Users mailing list
> >        > To UNSUBSCRIBE or update options visit:
> >        >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >        >
> >        _______________________________________________
> >        --Bandwidth and Colocation provided by Easynews.com --
> >
> >        Asterisk-Users mailing list
> >        To UNSUBSCRIBE or update options visit:
> >           http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >        _______________________________________________
> >        --Bandwidth and Colocation provided by Easynews.com --
> >
> >        Asterisk-Users mailing list
> >        To UNSUBSCRIBE or update options visit:
> >           http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> >
> >
> >
> > --
> > Mike
> > Sales Manager
> > http://www.theclubvoip.com
> > Making it happen
> > 1.888.470.7253
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>